Purpose: This tutorial is intended to help provide a new or intermediate user with a comprehensive walkthrough that will allow them set up an account with Voxalot for VOIP telephone calls using a real telephone handset.

(While Voxalot can be configured to work with any SIP device and any bring-your-own-device (BYOD) SIP provider, this walkthrough is geared toward configuring a Linksys Sipura telephone adapter using providers tested and recommended by the author.)

This walkthrough will guide you through the following steps so that you can enjoy the savings that VOIP telephony now offers to people:
-Creating a Voxalot account
-Purchasing an ATA online
-Configuring an ATA, with specific recommended settings for a Sipura ATA
-Purchasing an inbound phone number and outbound calling minutes
-Configuring your Voxalot account in your Voxalot Member’s Portal to use your new calling services

1. Creating a Voxalot Account

You will need a Voxalot account to get anywhere. If you already have one, good. If not, read on. At the Voxalot homepage, click on "Register" on the right-hand column, and at the next page create a six digit user name for your new account, enter and re-enter a password for this account, and provide a valid email address. Submit that, and you will get an email at the address you listed that will give you a link to activate your account. Once you've clicked that link, your account is ready for use and will be the basis to support the rest of the features discussed in this guide.

Login to your member page, and input the country you are in, and choose the closest server cluster to you; if you're in North America, then pick us.voxalot.com, if you're in Europe, choose eu.voxalot.com, and if you're in Australia, choose au.voxalot.com. If you're not quite in any of those places, pick the one that sounds closest geographically. (You can ping each url to get an idea as well what server is the closest.) Then pick the timezone you're in, and whether or not you want to have voice mail service with your number. I recommend you keep it enabled, it will come in handy after we set up your account to mimic a real telephone service. (Most local phone carriers charge you extra for Voice Mail while with Voxalot it is free.) If you have enabled voice mail, then select a ringing duration and a four-digit numeric password that suits your preferences.

(The "Enable symmetric NAT handling" setting should ideally be set to No--it provides outbound proxying service if your connection to the open internet proves difficult to manage using STUN alone; this setting at Yes basically helps some people in some cases not have audio problems, at the cost of increasing the delay between the time the words are spoken and the time the other side hears the words. Come back later and see if you can change this setting to "No" and keep your configuration working. Otherwise you need to keep it at yes.)

Click save, and you're done here for now.

2. Purchasing a Telephone Adapter Online

You're going to need to buy a piece of equipment now if you want to use VOIP with your regular telephone handset. It is a device called an ATA (analog telephone adapter) that you plug into your router that goes into your cable modem, and lets you plug an ordinary phone into it to simulate very convincingly a traditional telephone connection to the phone company. If you already have an unlocked adapter you can move on to the next topic.

There are many places to purchases this equipment online, especially if you are fortunate enough to live inside the United States. This is a Canadian store where I bought my equipment from: NCIX.com - Buy VOIP / Internet Phone In Canada They should sell exactly what you need for prices that are pretty good. However there are many other options if you live in a different country or are in search of a deal, this is a good community site that lists them: Cheapest ATAs and Service - voip-info.org

I am going to recommend you buy a Sipura ATA over a Grandstream model. They are high(er) quality, they are eminently configurable and customizable, widely respected and it is the adapter that I am using that I will give tips for configuring. In particular I recommend buying the SPA-1001, the PAP2T-NA, the SPA-2102 or the SPA-3102. I am also a big fan of the Sipura/Linksys IP telephones, e.g. the SPA941/942 or the SPA962. Try to purchase a device that still is having its firmware actively updated and maintained by Cisco/Linksys. Cisco is constantly dropping support for some of these devices and then introducing new ones so it will take some work to stay on top of it.

A Grandstream ATA or IP phone user can accomplish most of the following as well, and offers other codecs including iLBC that is more compatible with people on dial-up connections who are using free softphones like Gizmo5 or X-lite that do not support G729 or G723.1 (sigh... patents). However I won't vouch for their equipment's rock solid quality the way I can with Sipuras. Grandstream devices seem a little cheaper and less feature-rich, despite offering more codecs which its big plus. (I can't wait for Skype's SILK ultra-widebrand codec to be rolled out to all our SIP VoIP devices!)

Place your order for the equipment and wait eagerly for it to come in the mail. It will cost you about C$60 to C$120, or less, after taxes and shipping and handling costs are factored in, depending on the ATA/IP phone you selected and features it has. But the savings in your monthly phone bill can very quickly repay this one-time investment in equipment.

3. Configuring your ATA

A. When you get your Sipura, if you purchased a SPA-1001 or a PAP2T-NA, you will plug the ethernet cable it came with into both it and your router. Plug in the AC power adapter that came with it as well.

B. If you selected a SPA-2102 or SPA-3102, you should power it up and plug it directly into your PC's network card temporarily (i.e. unplugging your PC from the router and plugging it directly into the ethernet port of the SPA-2102/3102 labelled as the "PC" port). Your LAN connection will reset (advanced users can speed it up by an ipconfig /release, then ipconfig /renew) and you can now log into your Sipura. (You can find out the address you need to type into your browser by clicking Start, then Run... then in the window that pops up type 'cmd' without the quotes, then in the new DOS prompt window that comes up type "ipconfig" without the quotes and see what what is said under the "Default Gateway . . . . . . . . . : 192.168.1.1" line. What is the IP address that it says there? It will be either 192.168.1.1 or something very close to it, like 192.168.2.1 or 192.168.0.1. Type that IP address into your web browser, it will take you to the Sipura's internal web administration page.) Go to the Sipura webpage, then in the upper right click "Admin Login" then click "Advanced". Now click the WAN tab. Under the Remote Management option, at "Enable WAN Web Server" select "Yes" (change it from no). Click Submit All Changes. Now you can unplug your computer from the Sipura and plug your computer back into the router--as it was originally. Your PC's internet connection will shortly be restored to normal. Plug the ethernet cable that came with the Sipura into your router at one end and into the "WAN" labelled port now. From here on, the SPA-2102/3102 can be treated like a SPA-1001 or a PAP2T-NA, from the point of view of this tutorial.

We're now going to set up your Sipura to work with Voxalot. You have to find the IP address of the Sipura on your LAN, you'll need it to log in to the configuration page. (Advanced users can find the Sipura's LAN address by logging into their router's setup page and seeing what devices are connect to the LAN and what each of their IP addresses are.) The way to find it is to plug an ordinary phone into the "Line 1" or "VOIP" (for a SPA-3102) telephone port of the Sipura, then lifting up the handset and dialing star four times,
i.e. **** (star-star-star-star).

It will read fairly quickly a set of numbers, starting with 192.168. Try to copy it down or memorize it. You can dial 110# again as many times as you need till you have the whole address down. Type that whole address in your webbrower (literally right in the address bar of Internet Explorer or Firefox) and press enter,
ex. 192.168.0.103

Now the Sipura web administration page will load up in your browser. If you have either the SPA2102/3102, click the "Voice" tab at the very top of the page. SPA-1001 or PAP2T-NA owners will already be at that page. Now click "Admin Login" in the upper-right then click "Advanced". Bookmark this page in your web browser now, you will come back here in the future a lot. When I say in the future go to your Sipura admin page, I mean go to this bookmark.

(This might be a good time to update your firmware for your Sipura, and your NAT router as well, if you are comfortable with either operation. The latest firmware is supposed to be the least buggy and most refined programming for your Sipura and your router available at present, and it is probably wise to update it so that your VOIP experience is less plagued with frustration. With a Sipura, the website Welcome to Linksys.com contains the latest firmware, under Downloads, then the Voice over IP (VoIP) category, then VOIP Adapters, then pick the model you have, and select the version of your adapter and see if the newest firmware they offer is newer/more recent than the one reported on your Sipura admin page Info tab, under Product Information--Software Version, and if so, proceed to update your adapter.)

Here are the setting I am asking you set up right now:

Under the System tab (SPA-1001/PAP2T-NA users only; found elsewhere in the 2102/3102):

Under the SIP Parameters heading:
Use Compact Header: no (setting this to YES is an optional luxury setting that may cause unexpected problems but that can reduce the amount of bandwidth used by your Sipura at potentially no loss of quality or performance)

Under the RTP Parameters heading:
RTP Packet Size: 0.020 (this setting should be set to 0.020 for typical optimized performance; it should be set at 0.030 if you are using the G723 codec as your preferred codec or have a slow connection or otherwise want to save as much bandwidth as possible at the loss of some audio quality; you will experience incompatibility with some providers by setting this to 0.010 so I keep mine at 0.020)
No UDP Checksum: no (setting this to YES is an optional luxury setting that may cause unexpected problems but that can reduce the amount of bandwidth used by your Sipura at potentially no loss of quality or performance, however default is usually BETTER)

NAT Support Parameters heading (my thanks to "boatman" for doing the hard work of learning what are the ideal settings):
Handle VIA received: Yes
Handle VIA rport: Yes
Insert VIA received: Yes
Insert VIA rport: Yes
Substitute VIA Addr: Yes
Send Resp To Src Port: Yes
STUN Enable: Yes
STUN Test Enable: Yes (contrary to widespread advice on the Internet, 'yes' appears to be the correct choice for people who do not know what kind of router they have and whether it is SIP compatible; however they do know it is non-symmetric type router, then this should be set to No to conserve bandwidth)
STUN Server: stun.voxalot.com (different servers can be found at STUN - voip-info.org under "Public STUN servers", if you're phone spontaneously stops working and stops registering, it might be worth it to change this STUN server to a different one. This has immediately cured random inbound and outbound calling problems I've had in the past.)
EXT IP: (blank)
EXT RTP Port Min: (blank)
NAT Keep Alive Intvl: 59 (may need to be set somewhat lower if you have an commercial-class router, using 59 as I seem to have to use on my pfSense FreeBSD-based router under 'normal' firewall optimization. In principle, you could try using values such as 29, 59, 89, 119, 149 and 179. The first thing you will notice if you set it too high is that your inbound calls no longer ring your device, while outbound calls appear to work fine. But setting it too low wastes bandwidth for both your connection and the Voxalot server.)

Under the Provisioning tab:

Under the Configuration Profile heading:
Provision Enable: No

Under the Regional tab:

Under the Control Timer Values (sec) heading:
VMWI Refresh Intvl: 300

Under the Miscellaneous heading:
Time Zone: (pick the one you're in)
Daylight Saving Time Rule: start=3/8/7/2:00;end=11/1/7/2:00;save=1 (if you're in the US or Canada otherwise do a Google search for your Australian etc. variant)
FXS Port Input Gain: -3 (note this is the default value: turn up or down in increments of three [e.g. ...,-9, -6, -3, 0, 3, 6, 9,...]; increase if the other person can't hear you well, decrease if you hear your own echo)
FXS Port Output Gain: -3 (note this is the default value: turn up or down in increments of three [e.g. ...,-9, -6, -3, 0, 3, 6, 9,...]; increase if you can't hear the other person well, decrease if the other person reports hearing their own echo)

Under the Line 1 tab (SPA-1001/2102/PAP2T-NA) or VOIP tab (SPA-3102):

Under the Network Settings heading:
Network Jitter Level: high (default is High; you may wish to experiment with Medium or Low in order to reduce latency in your phone calls)
Jitter Buffer Adjustment: up and down (default is up and down; you may wish to experiment with down only, or disable to reduce your ATA over-aggressively increasing the buffer--and phone latency--and increase lag)

Under the Subscriber Information heading:
Display Name: 2125551234 (choose your real phone number you want people you call to see that some providers pass along, so that if your real phone number was +1-212-555-1234, you should enter 2125551234 here)
User ID: 123456 (write your Voxalot account number in this space)
Password: password (write your Voxalot account password in this space)
Use Auth ID: Yes
Auth ID: 123456 (write your Voxalot account number in this space)

Under the Audio Configuration heading:
Preferred Codec: G711u (choose G711u if you are in North America and have at least 256kbps upload, choose G711a if you are in Europe or Australia and have at least 256kbps upload, choose G729a if you have below 256kbps upload capacity)
Echo Supp Enable: No (this is optional; Echo Suppression is the more primitive solution to telephone echo and has potential negative side-effects that Echo Cancellation does not have--I turn Echo Supp off normally without issue)
Leave everything else as is

For American or Canadian users who are accustomed to dialing 7-digits, use the following but change "604" in the following to your local area code:(*[56]00S0 |*xxx[x*]. |x |*xx |**x. |1[89]76x.! |1[79]00x.! |1[2-9]xx[2-9]xxxxxxS0 |#x1[2-9]xx[2-9]xxxxxxS0 |011xx. |411S0 |<:1604>[2-9]xxxxxxS0 |#x<:1604>[2-9]xxxxxxS0 )
For American and Canadian users who are accustomed to dialing 10-digits for local phone calls, use the following but change "604" and "778" to the respective local area codes you know:(*[56]00S0 |*xxx[x*]. |x |*xx |**x. |1[89]76x.! |1[79]00x.! |1[2-9]xx[2-9]xxxxxxS0 |#x1[2-9]xx[2-9]xxxxxxS0 |011xx. |411S0 |<:1>604[2-9]xxxxxxS0 |#x<:1>604[2-9]xxxxxxS0 |<:1>778[2-9]xxxxxxS0 |#x<:1>778[2-9]xxxxxxS0 )
For all other users, you're on your own! I am pretty unfamiliar with other national dialing conventions. Please help contribute to this tutorial by suggesting dial plans that fit other national conventions. Also, the above dialing plans are fairly complicated, and can be safely shortened to the default Voxalot Tutorial one without appreciable loss in function.

Click "Submit All Changes"! (Most important step, otherwise all these changes will not have stuck and they'll have to be re-entered.)

That's it, now you can test your phone.

Pick up the phone and dial *500. (Star-five-zero-zero.)

Do you get connected to your voice mail? Good. Enter your password you set at your member's page. Does it work? Good. (If not, you'll have to send a message to the Voxalot Forums and we might be able to help you.)

Hang up, pick up the phone again and dial *600 (Star-six-zero-zero.)

Do you get the Echo Test voice answering? Good. When the voice is done speaking talk into the phone, do you hear your voice repeated back to you with a short delay? Good. Outbound calling is working fine. Now everything is set up in your Sipura for the moment.

4. Purchasing VOIP Account Services:

I am assuming you want to get full calling features now to simulate a real telephone service. If not you can decide whether any of the following instructions are suitable for you or not. Though normally, receiving calls from regular phones and making calls to regular phones is a service provided by the same telephone provider, I've personally found it more thrifty and economic when it comes to VOIP, to shop around for services and consider inbound and outbound calling as two distinct categories that we can find different providers for, if that's what makes the most sense. First I am going to covering purchasing service for outbound (i.e. calling out) service to the regular phone network, and then I will cover aquiring service for inbound service (i.e. calling in or receiving calls) to the regular phone network.

A. Outbound Provider Recommendations

I am going to describe three telephone providers that I've used, tested alone and with the Voxalot service, and can recommend to people freely.

This is a good provider in many ways. It is reliable, good quality, good support, and fairly affordable. It offers three main phone plans to consider:
1) Pay-as-you-go: 2 cents/minute to Canada/United States and low international rates. Minimum purchase of $5.00 is required to begin service with this calling plan. For most people this plan makes more sense than the next two.
2) North America Unlimited: $19.95/month for calling the US & Canada. Logically, you should be talking to North Americans for at least 1000 minutes a month (every month) before this plan makes more sense to consider.
3) World Select: $29.95/month for calling many countries in the world.

The procedure for signing up with them involves clicking the Sign Up link, entering the information asked for, and then receiving an email that contains your "Web Login" username and password and your "Phone/SIP Login" username and password, which are different from each other. Your account by default is the free version which has limited features and no outbound calling to regular phones. The "Phone/SIP Login" info will be used later in configuring this provider for Voxalot. The "Web Login" info is what is needed to actually buy a calling plan. Using the "Web Login" info, log in to through the Callcentric webpage to your account portal. Once logged in, the way to purchase one of the above calling plans is the link under "My Products" which says "View details and modify your products here". From there it is fairly straightforward to select a plan, choose your product (in our case, you should say you will bring or use your own SIP device), enter your credit card number and make your purchase, or use Paypal.

Description: Callcentric
Username: 1777xxxxxxx (the username given in your "Phone/SIP Login" info)
Password: xxxxxxx (the password given in your "Phone/SIP Login" info)
Confirm Password: xxxxxxx (the password given in your "Phone/SIP Login" info)
Codecs: ulaw;alaw;g726;g729;ilbc;gsm (enter this if you have greater than or equal to 256kbps uplink and live in N. America)alaw;ulaw;g726;g729;ilbc;gsm (enter this if you have greater than or equal to 256kbps uplink and live other than in N. America)g729;ilbc;gsm;g726;ulaw;alaw (enter this if you have less than 256kbps uplink)
Host: callcentric.com
Port: 5060
Active: Yes
SIP Register: No (enter "Yes" if you purchased any plans that include inbound phone numbers with them or purchased one separately)
Click "Save"

This is also a good provider in many ways, sharing the same virtues as Callcentric but offering one additional compelling feature: outbound caller ID setting, so that when you call out with this provider, the people you call will see on their call display the number you entered at the Teliax account portal. Similarly to Callcentric, it offers three main plans:
1) Pay-as-you-go: 2 cents/minute to Canada/United States and low internatioanl rates. Minimum purchase of $10.00 is required to begin service with this plan.
2) Residential Unlimited: $24.99/month for unlimited US/Canada calling, includes an inbound USA local phone number with it.
3) Residential International Unlimited: $34.99/month for unlimited calling to many major countries, includes an inbound USA local or toll-free number with it.

To sign up with Teliax, click the Sign Up link, choose the plan you wish to purchase, enter all the information that is required and pay with your credit card or with Paypal. When it asks you to select hardware choose "I will use my own SIP device" if you have already purchased an ATA online. You will receive in email confirmation of your purchase and another email with your "Web Portal Info" username and password and your "SIP Device Info" username and password. You'll need the "SIP Device Info" to configure your Voxalot account to use Teliax. When you first log in to the web account portal you may have to activate your account with a number they sent to you by email for activation. Logging in is also a good idea so you can set your outbound caller ID that people see when you call them via Teliax.

Description: Teliax
Username: user (the username given in your "SIP Device Info" info)
Password: xxxxxxx (the password given in your "SIP Device Info" info)
Confirm Password: xxxxxxx (the password given in your "SIP Device Info" info)
Codecs: ulaw;alaw;g726;g729;ilbc;gsm (enter this if you have greater than or equal to 256kbps uplink and live in N. America)alaw;ulaw;g726;g729;ilbc;gsm (enter this if you have greater than or equal to 256kbps uplink and live other than in N. America)g729;ilbc;gsm;g726;ulaw;alaw (enter this if you have less than 256kbps uplink)
Host: voip-co1.teliax.com (enter the proxy listed in your Teliax account portal under the "Support" tab; may be different)
Port: 5060
Active: Yes
SIP Register: No (enter "Yes" if you purchased any plans that include inbound phone numbers with them or purchased one separately)
Click "Save"

This is also a good provider, which seems reliable and high quality. They offer only one pay-as-you-go plan but in many ways it is a great deal.
1) Pay-as-you-go: 1.39 cents/minute to Canada/United States and low internatioanl rates. Minimum purchase of $35.00 is required to begin service. Permits outbound caller ID setting by entering a 10-digit US or Canadian phone number in the "Display Name" portion of your Sipura (under "Line 1" or "VOIP", under the Subscriber Information heading), which will show up on the call display of people you call. You would have to call for over 1400 minutes a month outbound to the US & Canada before the Callcentric or Teliax plans became more cost-effective.

To sign up, click the "Click Here" part of "Click here to sign up today!", enter the requisite information, and credit card details and make your purchase. You will be sent an email with your username and password, which in the case of Vitelity, is the same information for both logging into your web account portal as well as using a SIP device.

Description: Vitelity Outbound
Username: user (the username given in your "SIP Device Info" info)
Password: xxxxxxx (the password given in your "SIP Device Info" info)
Confirm Password: xxxxxxx (the password given in your "SIP Device Info" info)
Codecs: ulaw;alaw;g726;g729;ilbc;gsm (enter this if you have greater than or equal to 256kbps uplink and live in N. America)alaw;ulaw;g726;g729;ilbc;gsm (enter this if you have greater than or equal to 256kbps uplink and live other than in N. America)g729;ilbc;gsm;g726;ulaw;alaw (enter this if you have less than 256kbps uplink)
Host: outbound1.vitelity.net (enter the Proxy listed at your Vitelity account portal, under the "Support" tab, at the category "Asterisk Configuration Samples --> SIP.conf" --> "[vitel-outbound]" --> "host=outbound1.vitelity.net"; your one might be slightly different from the one given here as an example)
Port: 5060
Active: Yes
SIP Register: No
Click "Save"

**Tip: I had hit-and-miss results getting my outbound caller ID to show the number I set in my "Display Name" calling this way. There is a more certain way to do this however. In the Vitelity account portal, under the "Sub Accounts" tab, you can create another SIP username and password for your Vitelity account, that you can force a certain outbound caller ID to show up on the calls you place. This way works well. You can keep using the above settings, but change the username and password to the "sub-account's" username & password instead of your "main account's" username and password.

B. Inbound Provider Recommendations

I am going to first recommend a company the specializes in providing phone numbers but charges a fairly high price. Then I'll share with you two companies that provide you with free American phone numbers (albeit to isolated rural communities in sparsely populated rate-centers). Then I'll remind you that Callcentric, Teliax and Vitelity also sell phone numbers that you should consider.

THIS COMPANY IS REMOVING THIS OPTION STARTING APRIL 1, 2010. LOOK ELSEWHERE FOR DID. This company offers phone numbers in most countries and typically for all the major cities in the countries they do support, and often many of the less important cities and towns as well. They are high quality, reliable and trustworthy with excellent tech support. The procedure for signing up with them is to first register a (free) account at the website using a username and password you select, then activate your account from the email they send you. Then log in to your account portal using the username and password you chose, and click on "Purchase" then "DIDs". You will see a country list, choose the country and city you need. Select your quantity (typically just "1") and add to cart. Then you can check out. (You can see the price of these numbers either as you are checking out, or also via clicking "Price List" then "DIDs" at the top.) You can make your purchase with credit card or Paypal. Since it is unlimited incoming, they charge a stiff price for the number. I'd look elsewhere unless it beats the price point of the Callcentric/Vitelity options. THIS COMPANY IS REMOVING THIS OPTION STARTING APRIL 1, 2010. LOOK ELSEWHERE FOR DID.

To configure it for Voxalot do the following. Click on "Configure" then "URIs". Under 'Add A New URI:'
protocol: SIP
new URI: usernumber@us.voxalot.com (where usernumber is your Voxalot user id, and us.voxalot.com is the server you are registering your Sipura with, which is either us.voxalot.com or eu.voxalot.com or au.voxalot.com)
click "create"

Now click "Configure" and then "DIDs and Trunks".
In the first screen, next to your number, under 'Mapped To' is presently mapped to 'No forwarding'. Change this to the URI you just created (e.g. usernumber@us.voxalot.com). Click "go".

Next within that panel there is a set of tabs under the words "Configuration", marked DID, POP, and DNS. DID is already selected. Now click POP. Here you can choose for your number between "US" and "BE" (and soon, "HK"). US means New York City, BE means Brussels Belgium, and HK means Hong Kong. Select the location that is geographically closest to you, so that the distance between you and the phone center is as close as possible. This will improve phone call quality and reduce latency. Click "go".

Now within that same set of tabs, click "DNS". Select "DNS SRV" for your phone number, because Voxalot takes advantage of this extra functionality. Click "go". Your number is ready.

I don't want to get too much into this one but this can provide you with a free USA phone number (actually located within rural Iowa) that you can forward to your Voxalot account. After signing up for this one, you can log in to the account portal and set up forwarding. Under "Call Forward to Phone 1:" write sip:usernumber@us.voxalot.com (or eu.voxalot.com or au.voxalot.com) and click "Save".

Each of these providers, once you are logged in, provides you with the option of adding a phone number (DID) to your account. They may have already included a phone number you chose if you selected a more expensive calling plan for yourself when or if you signed up. These are good alternatives or substitutes for Voxbone.

If you have any DID with these providers login to your Voxalot account portal. Under "Member Menu" click "Providers".

If you have a DID with Callcentric or Teliax, at the entry for that provider, click "Edit". Make the following change:
SIP Register: Yes (change it from No if not already done; "No" is the correct setting when you only call out through that provider)

If you have a DID with Vitelity, you will have to create a new provider entry:
Description: Vitelity Inbound
Username: user (the username either given in your "SIP Device Info" info or the username of your subaccount)
Password: xxxxxxx (the password given in your "SIP Device Info" info or your subaccount's password)
Confirm Password: xxxxxxx (the password given in your "SIP Device Info" info)
Codecs: ulaw;alaw;g726;g729;ilbc;gsm (enter this if you have greater than or equal to 256kbps uplink and live in N. America)alaw;ulaw;g726;g729;ilbc;gsm (enter this if you have greater than or equal to 256kbps uplink and live other than in N. America)g729;ilbc;gsm;g726;ulaw;alaw (enter this if you have less than 256kbps uplink)
Host: inbound1.vitelity.net (enter the Proxy listed at your Vitelity account portal, under the "Support" tab, at the category "Asterisk Configuration Samples --> SIP.conf" then the bolded portion of the first line "register => user:password@inbound1.vitelity.net:5060"; this might be different somehow from the one given here as an example)
Port: 5060
Active: Yes
SIP Register: Yes
Click "Save"

Note 2: If you created sub-accounts in Vitelity, you have to make sure the Vitelity account (main or sub-) your DID is being forwarded to is the one that is being registered with the Voxalot service, otherwise you would be telling Voxalot to answer calls at one Vitelity account while your Vitelity portal is saying that the number is being sent to a different Vitelity account. Click the "DIDs" link on the side of your Vitelity account portal to configure this properly.

5. Setting up a Voxalot Dial Plan

There is one more step to be done to let these provider services interact with Voxalot and your Sipura. Login to your Voxalot account portal. Under "Member Menu" click "Dial Plans".

Let's suppose you have only one provider, Vitelity, for outbound calling. At the Dial Plan screen this would be sufficient:

To set that up, click "Add" then priority enter a number, e.g. 40 or 99 [lower numbers supercede higher ones!], then enter the following:
If the number begins with | 011NX route to Vitelity

Then do the same for another entry but enter this:
If the number equals | 1NXXNXXXXXX route to Vitelity

At this point you are ready to go and your phone service is now completely configured. Test it to see.

However if you had two providers, say Callecentric and Vitelity, and you wanted International calls to be by default handled by Callcentric and North American calls to be by default handled by Vitelity then you could enter this plan:

This would entail the following, by default International calls go by Callcentric and North American calls by Vitelity, however when prefixing #1 (pound-one) before the phone number it would be routed through Callcentric even if it was a North American number, and when prefixing #2 (pound-two) before the number, it would be routed through Vitelity even if it was an international number. And it can get more complicated when you add in a third and fourth provider to the mix (my suggestion would be to add more 'pound-number' rules to the Voxalot dial plan so you can access these other providers by prefixing pound-three etc. before the number you wish to dial while still maintaining default providers you usually call with where you don't need to add any special prefixes to the mix).

To create rules 51 and 52 in the above example you would have to click "Add" then "Advanced Mode" and then, for instance, enter the following:

If there is some major audio or calling problem and you are sure you are using the above settings then see if the following helps:
Setup special firewall or port-forwarding rules in your router to "help" your VOIP by forwarding ports the RTP ports (typically UDP ports 16384 to 16482 by default in a Sipura, UDP ports 8000 to 8010 in X-lite, and UDP ports 5004-5007 in a Grandstream device) to your ATA, or by putting your ATA in your router's DMZ (which has the same effect but in essence forwards all ports to the ATA).

DMZing/port-forwarding would be the first place I would look in order to solve an audio problem. The reason why it might be necessary is because to communicate in Voice Over IP, you have to get a clean connection between your phone adapter and the outside internet, both incoming and outgoing. In some cases routers are trying to do their job and 'protect' you by dropping data from reaching your ATA before it has a chance to see it, because according to the NAT router's programming, it is an unsolicited 'attack' on your network from somewhere on the Internet. STUN tries to overcome this and often can by itself fix your connection, but in some cases because there are many different types of routers with different programming, you might find you need to add in additional port-forwarding or DMZing your ATA in order to stop your router from discarding packets before they reach you.

On some routers this fixes things. On others, doing this actually can interfere with the special STUN features already in use by the Sipura to compensate for NAT routing, and breaks a working configuration. Routers really behave differently with lots of variations.

After tweaking these settings it is worth unplugging the Sipura's power and plugging it back in so it reconfigures itself properly with the router and the internet. If this didn't solve your issue, now try the putting-the-Sipura-in-the-DMZ strategy again. If all this has failed, you should seriously consider testing with a different router. Some routers are just plain uncooperative with VOIP and the network techniques used to make phone calls; changing the router to a different one or enabling 'gaming mode' in the router will go a long way to solving some of the more difficult audio problems than can occur.

7. Touch-Tone (DTMF) Audio Problems

Is the interactive voice response system (IVR) you are calling not accepting your touch tones codes? It might either think you didn't enter any or it thinks you entered the wrong one, or thinks you entered it twice or more when you only entered it once. This is an annoyingly frequent problem in VOIP. There is no one answer to this problem. However the following works most of the time.

The best trick I have found to get touch-tones to work is to change DTMF Tx Method from Auto to either InBand or AVT or INFO. InBand frequently works the best, but on the other hand through some providers it doesn't work at all, and AVT or INFO seem to work. Alternately, you can keep DTMF Tx Method at Auto and toggle DTMF Process INFO and DTMF Process AVT from the present settings then try the four possible combinations,
yes, yes
yes, no
no, yes
no, no

This seems to have a similar effect to changing the DTMF Tx Method directly. I have nearly full reliability when I discover what settings the Provider I am using 'likes' for DTMF. However when you call out with a different provider, or even use the same provider to a different area code, or even the same provider and same number but after some time has passed, your 'working' configuration no longer works and you will need to re-tweak the above DTMF settings in the Sipura to get the IVR to understand you. It is unfortunate this is still a problem in VOIP (the Auto option is supposed to resolve this for you but it does not seem well-designed enough) but I wanted to give you tips on how to deal with it if you encounter touch-tone problems on IVRs.

8. Conclusion

This concludes my Voxalot tutorial. I hope it was of use to you in setting up your telephone system to use Voxalot in a way that simulates a traditional phone service but costs you much less. If you have any suggestions for improvement, corrections of mistakes, and particularly European and Australian examples for providers, and Sipura and Voxalot dial plans, let me know and I will incorporate any suggestions into this document with full credit.

I updated this post with new (corrected) NAT/STUN information -- thank you boatman, and I fixed the suggested dial plans which formerly didn't even let you use Voxalot's speed dial functionality (set up through the member portal). Any other suggestions, corrections, let me know while I am still in the editing frame of mind. Thanks.

Recently 'STUN Test Enable' has come to my attention. According to the Linksys ATA manual;

Quote:

If the STUN Enable feature is enabled and a valid STUN server is available, the Linksys ATA can perform a NAT-type discovery operation when it powers on. It contacts the configured STUN server, and the result of the discovery is reported in a Warning header in all subsequent REGISTER requests. If the Linksys ATA detects symmetric NAT or a symmetric firewall, NAT mapping is disabled.

The interesting part is "If the Linksys ATA detects symmetric NAT or a symmetric firewall, NAT mapping is disabled." I don't have a symmetric NAT router for testing, but it seems that "STUN Test Enable: yes" should work well with Voxalot's "Symmetric NAT Handling" option, which even when set 'yes' is not activated until NAT Mapping is disabled in the ATA. The ATA would then be portable between symmetric and non-symmetric routers without any configuration change. RTP packets would be proxied through Voxalot when necessary (when router is symmetric NAT) but when using a SIP compatible router RTP path would be direct for lower voice latency.

Some users may not know if they are behind symmetric NAT. Rather than asking the novice user to discover if they are behind symmetric NAT and manually turn off "NAT Mapping Enable" it might be better to have the ATA figure this out and automatically make the necessary adjustments.

Thanks great explanation! Voxalot's official instructions for the 3201 should refer here!

I want to forward '911' calls to my local police department eg 908 123 1234. Is the best place to do that; in your smart call list(dial plans) or in the ATA? What are the steps?

I'm not sure about the US, but in AU we are better off dialing the emergency number direct (000 in our case) -- the local police station may or may not be manned and furthermore emergency calls are a special case which could be routed by the call centre to police, fire or ambulance.

Wouldn't 911 be better off going to the "call centre' ?

Only non-urgent calls should be made to the "normal" local authorities, such as the police.

I was using 911 as an example - maybe bad 'cause of all the other issues it raises... Say i want 411 to convert to 800 Goog 411. Does somebody know what string to put into the Dial Plan: line? Is there documentation of that somewhere? I'm trying to avoid the complication of setting up a PBX just to route special numbers.

About emergency dialing - the US is a big place I used to live in NYC were the call center is important. Here in our town, all the emergency calls go into a local number so 911 in our case go to the local number that dispatches police fire ambulance. Most likely if we were calling, we'd be using one of the 4 mobile phones we have in the house...

On the 3102 Line 1 setup there is an "Emergency Number" field any idea what that does?

Quote:

Originally Posted by affinity

I'm not sure about the US, but in AU we are better off dialing the emergency number direct (000 in our case) -- the local police station may or may not be manned and furthermore emergency calls